Hi all After a long iam back to forum back to my own topic and several readings done on this forum how people doing same kind of setup what iam trying to achive
so here i have done some good developements for testing iam doing all in one Server Step1 : Installed in Fresh BOX with Debian Asterisk and A2B working Fine Step2 : registered with SIP account iam able to make calls successfully Step3 : installed Opensips Made Subscribers to view from A2b Database Step4 : changed Asterisk port from 5060 to 5062 Step5 : Opensip config made changes to register users with Opensips and when they dial 001X call send to Asterisk box route[3]{ if (uri =~ "sip:001[0...@*"){ log(1, "Forwarding to Asterisk \n"); rewritehostport("A2b-asterisk-IP:5062"); route(1); exit; } Works Fine, No problems as of now But to go in advance, i want to use Number of * boxes to achive more Load Step5 : added Dispatcher Module in the Opensips loadmodule "dispatcher.so" . . . modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg") . . . . changed route to use dispatcher route[3]{ if (uri =~ "sip:001[0...@*"){ log(1, "Forwarding to Asterisk \n"); ds_select_dst("2","4"); forward(); route(1); exit; } My dispatcher Config Looks like below dispatcher.cfg 2 sip:a2b-asterisk-ip:5062 2 sip:a2b-asterisk-ip2:5062 I have restarted Opensips when i dial 0017XXXXXX number the call send Opensips to Asterisk Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2] Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1] Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected [4-2/1] <sip:a2b-asterisk-ip:5062> Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS lookup... Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request: sending:#012INVITE sip:0017xxxxx...@opensips-ip:5060 SIP/2.0#015#012Record-Route: <sip:opensips-ip;lr=on>#015#012Via: SIP/2.0/UDP opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From: 4720779942 <sip:4720779...@opensips-ip:5060>;tag=1966722825#015#012To: 0017325824631 <sip:0017xxxx...@opensips-ip:5060>#015#012Call-ID: 32167199575863-11502744529...@ip-phoneip#015#012cseq: 2 INVITE#015#012Contact: <sip:4720779...@ipphone-ip:5060>#015#012Proxy-Authorization: Digest username="4720779942", realm="asterisk", nonce="79ee65ba", uri="sip:0017xxx...@opensips-ip:5060", response="3e182f165a5663d0b145d6b55d34e94b", algorithm=MD5#015#012Max-Forwards: 69#015#012Supported: replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE#015#012Content-Type: application/sdp#015#012Content-Length: 319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4 202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0 0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9 G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012. opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220, proto=1 when i ngrep ------------ U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060 INVITE sip:0017xxxxx...@asterisk-a2b-ip:5060 SIP/2.0. Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport. From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266. To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>. Call-ID: 14399316162240-7371067914...@ipphone. CSeq: 2 INVITE. Contact: <sip:4720779...@ipphone:5060>. Proxy-Authorization: Digest username="4720779942", realm="asterisk", nonce="07ba8624", uri="sip:0017xxxxx...@asterisk-a2b-ip:5060", response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5. Max-Forwards: 70. Supported: replaces. User-Agent: Voip Phone 1.0. Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE. Content-Type: application/sdp. Content-Length: 319. . v=0. o=4720779942 69102627 18481147 IN IP4 ipphone. s=A conversation. c=IN IP4 ipphone. t=0 0. m=audio 10034 RTP/AVP 18 4 8 0 9 101. a=rtpmap:18 G729/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:9 G722/16000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=sendrecv. U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060. From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266. To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>. Call-ID: 14399316162240-7371067914...@ipphone. CSeq: 2 INVITE. Server: OpenSIPS (1.5.1-notls (i386/linux)). Content-Length: 0. . U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060. From: 4720779942 <sip:4720779...@asterisk-a2b-ip:5060>;tag=3037030266. To: 0017XXXXXXXX <sip:0017xxxxx...@asterisk-a2b-ip:5060>;tag=as0cb075c5. Call-ID: 14399316162240-7371067914...@ipphone. CSeq: 1 INVITE. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07ba8624". Content-Length: 0. ------ when i enable debug at Asterisk and Look at i see the below error --------------------------------------------------------------- <--- SIP read from a2b-asterisk-ip:5060 ---> INVITE sip:0017xxxxxx...@a2b-asterisk-ip:5060 SIP/2.0 Record-Route: <sip:a2b-asterisk-ip;lr=on> Via: SIP/2.0/UDP a2b-asterisk-ip;branch=z9hG4bK166.1b7e2827.0 Via: SIP/2.0/UDP Ip-phone:5060;received=Ip-phone;branch=z9hG4bK295731884823024293;rport=5060 From: 4720779942 <sip:4720779...@a2b-asterisk-ip:5060>;tag=12544334 To: 0017XXXXXXXXX <sip:0017xxxxxx...@a2b-asterisk-ip:5060> Call-ID: 16946271051109-143302828620...@ip-phone CSeq: 1 INVITE Contact: <sip:4720779...@ip-phone:5060> Max-Forwards: 69 Supported: replaces User-Agent: Voip Phone 1.0 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, UPDATE Content-Type: application/sdp Content-Length: 319 v=0 o=4720779942 31008195 22123120 IN IP4 Ip-phone s=A conversation c=IN IP4 Ip-phone t=0 0 m=audio 10030 RTP/AVP 18 4 8 0 9 101 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <-------------> [Jun 30 01:15:29] VERBOSE[24612] logger.c: --- (15 headers 14 lines) --- [Jun 30 01:15:29] VERBOSE[24612] logger.c: Ignoring this INVITE request [Jun 30 01:15:31] VERBOSE[24612] logger.c: Reliably Transmitting (no NAT) to termination-provider-ip:5062: OPTIONS sip:termination-provider-ip:5062 SIP/2.0 Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8 To: <sip:termination-provider-ip:5062> Contact: <sip:aster...@a2b-asterisk-ip:5062> Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 30 Jun 2009 08:15:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jun 30 01:15:32] VERBOSE[24612] logger.c: <--- SIP read from termination-provider-ip:5062 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP a2b-asterisk-ip:5062;branch=z9hG4bK6a9fe793;rport=5062 From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8 To: <sip:termination-provider-ip:5062>;tag=2560d490c3265ff35995c6bbde62a7c3.ee5a Call-ID: 65a49c0977c6de0a1d2dbbfe75772...@a2b-asterisk-ip CSeq: 102 OPTIONS Content-Length: 0 --------- why does Asterisk sending with out any values --- From: "asterisk" <sip:aster...@a2b-asterisk-ip:5062>;tag=as4cf91fd8 To: <sip:termination-provider-ip:5062> --- Any suggestions Ram
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